├── README.md
├── g711a.pcap
├── invite-accounts.csv
├── invite-auth.xml
├── invite-without-auth-expected-to-receive-busy-486.xml
├── invite-without-auth.xml
├── register-accounts.csv
├── register.xml
└── uas.xml


/README.md:
--------------------------------------------------------------------------------
 1 | # sipp2freeswitch
 2 | sample sipp scenarios for testing freeswitch
 3 | 
 4 | Used following resources:
 5 | 
 6 | https://github.com/jrzondagh/SIPp-Scenario-Files
 7 | 
 8 | https://github.com/saghul/sipp-scenarios
 9 | 
10 | ./sipp -i local_ip_address -sf ./register.xml -inf ./register-accounts.csv destination_ip_address:5060 -r 1 -rp 1000 -aa -trace_err
11 | 
12 | ./sipp -i local_ip_address -sf ./invite-auth.xml -inf ./invite-accounts.csv destination_ip_address:5060 -r 1 -rp 1000 -trace_err
13 | 
14 | ./sipp -i local_ip_address -sf ./invite-without-auth.xml -inf ./invite-accounts.csv destination_ip_address:5060 -r 1 -rp 1000 -trace_err
15 | 
16 | ./sipp -sf ./uas.xml -i local_ip_address -p 5060
17 | 


--------------------------------------------------------------------------------
/g711a.pcap:
--------------------------------------------------------------------------------
https://raw.githubusercontent.com/os11k/sipp2freeswitch/9c5d1b5e4b2347ba31f93a1805966cb9e6ca2656/g711a.pcap


--------------------------------------------------------------------------------
/invite-accounts.csv:
--------------------------------------------------------------------------------
 1 | SEQUENTIAL
 2 | # Username: 1000-1009
 3 | # Password: 1234
 4 | # SIP Proxy: 10.0.0.10, Destination Number: 9196
 5 | 1000;10.0.0.10;[authentication username=1000 password=1234];9196;
 6 | 1001;10.0.0.10;[authentication username=1001 password=1234];9196;
 7 | 1002;10.0.0.10;[authentication username=1002 password=1234];9196;
 8 | 1003;10.0.0.10;[authentication username=1003 password=1234];9196;
 9 | 1004;10.0.0.10;[authentication username=1004 password=1234];9196;
10 | 1005;10.0.0.10;[authentication username=1005 password=1234];9196;
11 | 1006;10.0.0.10;[authentication username=1006 password=1234];9196;
12 | 1007;10.0.0.10;[authentication username=1007 password=1234];9196;
13 | 1008;10.0.0.10;[authentication username=1008 password=1234];9196;
14 | 1009;10.0.0.10;[authentication username=1009 password=1234];9196;
15 | 


--------------------------------------------------------------------------------
/invite-auth.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="ISO-8859-1" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | 
  4 | <!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->
  5 | 
  6 | <!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
  7 | <!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
  8 | <!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->
  9 | 
 10 | <!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->
 11 | 
 12 | <scenario name="UAC INVITE with Auth and SDP">
 13 |   <send retrans="500" start_txn="invite">
 14 |     <![CDATA[
 15 | 
 16 |       INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
 17 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 18 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
 19 |       To: sut <sip:[field3]@[field1]:[remote_port]>
 20 |       Call-ID: [call_id]
 21 |       CSeq: [cseq] INVITE
 22 |       Contact: <sip:sipp@[local_ip]:[local_port]>
 23 |       Max-Forwards: 70
 24 |       Subject: Performance Test
 25 |       User-Agent: SIPp Tester UAC
 26 |       Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
 27 |       Content-Type: application/sdp
 28 |       Content-Length: [len]
 29 | 
 30 |       v=0
 31 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 32 |       s=-
 33 |       c=IN IP[media_ip_type] [media_ip]
 34 |       t=0 0
 35 |       m=audio [media_port] RTP/AVP 8
 36 |       a=rtpmap:8 PCMA/8000
 37 |       a=rtpmap:101 telephone-event/8000
 38 |       a=fmtp:101 0-11,16
 39 | 
 40 | 
 41 |     ]]>
 42 |   </send>
 43 | 
 44 |   <recv response="100" optional="true" rrs="true" response_txn="invite">
 45 |   </recv>
 46 | 
 47 |   <recv response="407" auth="true" rrs="true" response_txn="invite">
 48 |   </recv>
 49 | 
 50 |   <send ack_txn="invite">
 51 |     <![CDATA[
 52 | 
 53 |       ACK sip:[field3]@[field1] SIP/2.0
 54 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
 55 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
 56 |       [last_To]
 57 |       [routes]
 58 |       Call-ID: [call_id]
 59 |       CSeq: [cseq] ACK
 60 |       Contact: <sip:sipp@[local_ip]:[local_port]>
 61 |       Max-Forwards: 70
 62 |       Subject: Performance Test
 63 |       User-Agent: SIPp Tester UAC
 64 |       Content-Length: 0
 65 | 
 66 |     ]]>
 67 |   </send>
 68 | 
 69 |   <send retrans="500" start_txn="invite2">
 70 |     <![CDATA[
 71 | 
 72 |       INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
 73 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 74 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
 75 |       To: sut <sip:[field3]@[field1]:[remote_port]>
 76 |       [routes]
 77 |       Call-ID: [call_id]
 78 |       CSeq: [cseq] INVITE
 79 |       Contact: <sip:sipp@[local_ip]:[local_port]>
 80 |       Max-Forwards: 70
 81 |       Subject: Performance Test
 82 |       User-Agent: SIPp Tester UAC   
 83 |       Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
 84 |       [field2]
 85 |       Content-Type: application/sdp
 86 |       Content-Length: [len]
 87 | 
 88 |       v=0
 89 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 90 |       s=-
 91 |       c=IN IP[media_ip_type] [media_ip]
 92 |       t=0 0
 93 |       m=audio [media_port] RTP/AVP 8
 94 |       a=rtpmap:8 PCMA/8000
 95 |       a=rtpmap:101 telephone-event/8000
 96 |       a=fmtp:101 0-11,16
 97 | 
 98 | 
 99 |     ]]>
100 |   </send>
101 | 
102 |   <recv response="100" optional="true" response_txn="invite2">
103 |   </recv>
104 | 
105 |   <recv response="180" optional="true" response_txn="invite2">
106 |   </recv>
107 | 
108 |   <recv response="183" optional="true" response_txn="invite2">
109 |   </recv>
110 | 
111 |   <!-- Grab the 200 OK's Contact Header's URI for use in the ACK's Request URI -->
112 |   <recv response="200" rtd="true" rrs="true" response_txn="invite2">
113 | 	<action>
114 | 		<ereg regexp= "sip:[^+>]+"  search_in="hdr" header="Contact:" assign_to="1" />
115 | 	</action>
116 |   </recv>
117 | 
118 | 
119 |   <!-- decrement the ACK's branch by 3 to match previous INVITE's branch -->
120 |   <send ack_txn="invite2">
121 |     <![CDATA[
122 | 
123 |       ACK [$1] SIP/2.0
124 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
125 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
126 |       [last_To]
127 |       [routes]
128 |       Call-ID: [call_id]
129 |       CSeq: [cseq] ACK
130 |       Contact: <sip:sipp@[local_ip]:[local_port]>
131 |       Max-Forwards: 70
132 |       Subject: Performance Test
133 |       User-Agent: SIPp Tester UAC
134 |       Content-Length: 0
135 | 
136 |     ]]>
137 |   </send>
138 | 
139 | <nop>
140 |     <action>
141 |       <exec play_pcap_audio="./g711a.pcap"/>
142 |     </action>
143 | </nop>
144 | 
145 |   <pause milliseconds="8000"/>
146 |   
147 |   <send retrans="500">
148 |     <![CDATA[
149 | 
150 |       BYE [$1] SIP/2.0
151 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
152 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
153 |       [last_To]
154 |       [routes]
155 |       Call-ID: [call_id]
156 |       CSeq: [cseq] BYE
157 |       Contact: <sip:sipp@[local_ip]:[local_port]>
158 |       Max-Forwards: 70
159 |       Subject: Performance Test
160 |       User-Agent: SIPp Tester UAC
161 |       [field2]
162 |       Content-Length: 0
163 | 
164 |     ]]>
165 |   </send>
166 | 
167 |   <recv response="200" crlf="true">
168 |   </recv>
169 | 
170 |   <!-- definition of the response time repartition table (unit is ms)   -->
171 |   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
172 | 
173 |   <!-- definition of the call length repartition table (unit is ms)     -->
174 |   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
175 | 
176 | </scenario>
177 | 


--------------------------------------------------------------------------------
/invite-without-auth-expected-to-receive-busy-486.xml:
--------------------------------------------------------------------------------
 1 | <?xml version="1.0" encoding="ISO-8859-1" ?>
 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
 3 | 
 4 | <!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->
 5 | 
 6 | <!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
 7 | <!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
 8 | <!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->
 9 | 
10 | <!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->
11 | 
12 | <scenario name="UAC INVITE with Auth and SDP">
13 |   <send retrans="500" start_txn="invite">
14 |     <![CDATA[
15 | 
16 |       INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
17 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
18 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
19 |       To: sut <sip:[field3]@[field1]:[remote_port]>
20 |       Call-ID: [call_id]
21 |       CSeq: [cseq] INVITE
22 |       Contact: <sip:sipp@[local_ip]:[local_port]>
23 |       Max-Forwards: 70
24 |       Subject: Performance Test
25 |       User-Agent: SIPp Tester UAC
26 |       Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
27 |       Content-Type: application/sdp
28 |       Content-Length: [len]
29 | 
30 |       v=0
31 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
32 |       s=-
33 |       c=IN IP[media_ip_type] [media_ip]
34 |       t=0 0
35 |       m=audio [media_port] RTP/AVP 8
36 |       a=rtpmap:8 PCMA/8000
37 |       a=rtpmap:101 telephone-event/8000
38 |       a=fmtp:101 0-11,16
39 | 
40 | 
41 |     ]]>
42 |   </send>
43 | 
44 |   <recv response="100" optional="true" rrs="true" response_txn="invite">
45 |   </recv>
46 | 
47 |   <recv response="486" rrs="true" response_txn="invite">
48 |   </recv>
49 | 
50 |   <send ack_txn="invite">
51 |     <![CDATA[
52 | 
53 |       ACK sip:[field3]@[field1] SIP/2.0
54 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
55 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
56 |       [last_To]
57 |       [routes]
58 |       Call-ID: [call_id]
59 |       CSeq: [cseq] ACK
60 |       Contact: <sip:sipp@[local_ip]:[local_port]>
61 |       Max-Forwards: 70
62 |       Subject: Performance Test
63 |       User-Agent: SIPp Tester UAC
64 |       Content-Length: 0
65 | 
66 |     ]]>
67 |   </send>
68 | 
69 |   <!-- definition of the response time repartition table (unit is ms)   -->
70 |   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
71 | 
72 |   <!-- definition of the call length repartition table (unit is ms)     -->
73 |   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
74 | 
75 | </scenario>
76 | 


--------------------------------------------------------------------------------
/invite-without-auth.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="ISO-8859-1" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | 
  4 | <!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->
  5 | 
  6 | <!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
  7 | <!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
  8 | <!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->
  9 | 
 10 | <!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->
 11 | 
 12 | <scenario name="UAC INVITE with Auth and SDP">
 13 |   <send retrans="500" start_txn="invite">
 14 |     <![CDATA[
 15 | 
 16 |       INVITE sip:[field3]@[field1]:[remote_port] SIP/2.0
 17 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 18 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
 19 |       To: sut <sip:[field3]@[field1]:[remote_port]>
 20 |       Call-ID: [call_id]
 21 |       CSeq: [cseq] INVITE
 22 |       Contact: <sip:sipp@[local_ip]:[local_port]>
 23 |       Max-Forwards: 70
 24 |       Subject: Performance Test
 25 |       User-Agent: SIPp Tester UAC
 26 |       Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
 27 |       Content-Type: application/sdp
 28 |       Content-Length: [len]
 29 | 
 30 |       v=0
 31 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 32 |       s=-
 33 |       c=IN IP[media_ip_type] [media_ip]
 34 |       t=0 0
 35 |       m=audio [media_port] RTP/AVP 8
 36 |       a=rtpmap:8 PCMA/8000
 37 |       a=rtpmap:101 telephone-event/8000
 38 |       a=fmtp:101 0-11,16
 39 | 
 40 | 
 41 |     ]]>
 42 |   </send>
 43 | 
 44 |   <recv response="100" optional="true" response_txn="invite">
 45 |   </recv>
 46 | 
 47 |   <recv response="180" optional="true" response_txn="invite">
 48 |   </recv>
 49 | 
 50 |   <recv response="183" optional="true" response_txn="invite">
 51 |   </recv>
 52 | 
 53 |   <!-- Grab the 200 OK's Contact Header's URI for use in the ACK's Request URI -->
 54 |   <recv response="200" rtd="true" rrs="true" response_txn="invite">
 55 |         <action>
 56 |                 <ereg regexp= "sip:[^+>]+"  search_in="hdr" header="Contact:" assign_to="1" />
 57 |         </action>
 58 |   </recv>
 59 | 
 60 | 
 61 |   <!-- decrement the ACK's branch by 3 to match previous INVITE's branch -->
 62 |   <send ack_txn="invite">
 63 |     <![CDATA[
 64 | 
 65 |       ACK [$1] SIP/2.0
 66 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
 67 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
 68 |       [last_To]
 69 |       [routes]
 70 |       Call-ID: [call_id]
 71 |       CSeq: [cseq] ACK
 72 |       Contact: <sip:sipp@[local_ip]:[local_port]>
 73 |       Max-Forwards: 70
 74 |       Subject: Performance Test
 75 |       User-Agent: SIPp Tester UAC
 76 |       Content-Length: 0
 77 | 
 78 |     ]]>
 79 |   </send>
 80 | 
 81 | <nop>
 82 |     <action>
 83 |       <exec play_pcap_audio="./g711a.pcap"/>
 84 |     </action>
 85 | </nop>
 86 | 
 87 |   <pause milliseconds="8000"/>
 88 | 
 89 |   <send retrans="500">
 90 |     <![CDATA[
 91 | 
 92 |       BYE [$1] SIP/2.0
 93 |       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 94 |       From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid]SIPpTag00[call_number]
 95 |       [last_To]
 96 |       [routes]
 97 |       Call-ID: [call_id]
 98 |       CSeq: [cseq] BYE
 99 |       Contact: <sip:sipp@[local_ip]:[local_port]>
100 |       Max-Forwards: 70
101 |       Subject: Performance Test
102 |       User-Agent: SIPp Tester UAC
103 |       Content-Length: 0
104 | 
105 |     ]]>
106 |   </send>
107 | 
108 |   <recv response="200" crlf="true">
109 |   </recv>
110 | 
111 |   <!-- definition of the response time repartition table (unit is ms)   -->
112 |   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
113 | 
114 |   <!-- definition of the call length repartition table (unit is ms)     -->
115 |   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
116 | 
117 | </scenario>
118 | 
119 | 


--------------------------------------------------------------------------------
/register-accounts.csv:
--------------------------------------------------------------------------------
 1 | SEQUENTIAL
 2 | # Username: 1000-1009
 3 | # Password: 1234
 4 | # SIP Proxy: 10.0.0.10
 5 | 1000;10.0.0.10;[authentication username=1000 password=1234]
 6 | 1001;10.0.0.10;[authentication username=1001 password=1234]
 7 | 1002;10.0.0.10;[authentication username=1002 password=1234]
 8 | 1003;10.0.0.10;[authentication username=1003 password=1234]
 9 | 1004;10.0.0.10;[authentication username=1004 password=1234]
10 | 1005;10.0.0.10;[authentication username=1005 password=1234]
11 | 1006;10.0.0.10;[authentication username=1006 password=1234]
12 | 1007;10.0.0.10;[authentication username=1007 password=1234]
13 | 1008;10.0.0.10;[authentication username=1008 password=1234]
14 | 1009;10.0.0.10;[authentication username=1009 password=1234]
15 | 
16 | 


--------------------------------------------------------------------------------
/register.xml:
--------------------------------------------------------------------------------
 1 | <?xml version="1.0" encoding="ISO-8859-1" ?>
 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
 3 | 
 4 | <scenario name="registration">
 5 | 
 6 | <send retrans="500">
 7 | <![CDATA[
 8 | REGISTER sip:[field1] SIP/2.0
 9 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
10 | Max-Forwards: 70
11 | From: "sipp" <sip:[field0]@[field1]>;tag=[call_number]
12 | To: "sipp" <sip:[field0]@[field1]>
13 | Call-ID: reg///[call_id]
14 | CSeq: 7 REGISTER
15 | Contact: <sip:sipp@[local_ip]:[local_port]>
16 | Expires: 3600
17 | Content-Length: 0
18 | User-Agent: SIPp
19 | ]]>
20 | </send>
21 | 
22 | <recv response="100" optional="true">
23 | </recv>
24 | 
25 | <recv response="401" auth="true" rtd="true">
26 | </recv>
27 | 
28 | <send retrans="500">
29 | <![CDATA[
30 | REGISTER sip:[field1] SIP/2.0
31 | Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
32 | Max-Forwards: 70
33 | From: "sipp" <sip:[field0]@[field1]>;tag=[call_number]
34 | To: "sipp" <sip:[field0]@[field1]>
35 | Call-ID: reg///[call_id]
36 | CSeq: 8 REGISTER
37 | Contact: <sip:sipp@[local_ip]:[local_port]>
38 | Expires: 3600
39 | Content-Length: 0
40 | User-Agent: SIPp
41 | [field2]
42 | ]]>
43 | </send>
44 | 
45 | <recv response="100" optional="true">
46 | </recv>
47 | 
48 | <recv response="200">
49 | </recv>
50 | 
51 | <ResponseTimeRepartition value="10, 20"/>
52 | <CallLengthRepartition value="10"/>
53 | </scenario>
54 | 
55 | 


--------------------------------------------------------------------------------
/uas.xml:
--------------------------------------------------------------------------------
  1 | <?xml version="1.0" encoding="ISO-8859-1" ?>
  2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
  3 | 
  4 | <!-- This program is free software; you can redistribute it and/or      -->
  5 | <!-- modify it under the terms of the GNU General Public License as     -->
  6 | <!-- published by the Free Software Foundation; either version 2 of the -->
  7 | <!-- License, or (at your option) any later version.                    -->
  8 | <!--                                                                    -->
  9 | <!-- This program is distributed in the hope that it will be useful,    -->
 10 | <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
 11 | <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
 12 | <!-- GNU General Public License for more details.                       -->
 13 | <!--                                                                    -->
 14 | <!-- You should have received a copy of the GNU General Public License  -->
 15 | <!-- along with this program; if not, write to the                      -->
 16 | <!-- Free Software Foundation, Inc.,                                    -->
 17 | <!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
 18 | <!--                                                                    -->
 19 | <!--                 Sipp default 'uas' scenario.                       -->
 20 | <!--                                                                    -->
 21 | 
 22 | <scenario name="Basic UAS responder">
 23 |   <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
 24 |   <!-- are saved and used for following messages sent. Useful to test   -->
 25 |   <!-- against stateful SIP proxies/B2BUAs.                             -->
 26 |   <recv request="INVITE" crlf="true" rrs="true">
 27 |   </recv>
 28 | 
 29 |   <!-- The '[last_*]' keyword is replaced automatically by the          -->
 30 |   <!-- specified header if it was present in the last message received  -->
 31 |   <!-- (except if it was a retransmission). If the header was not       -->
 32 |   <!-- present or if no message has been received, the '[last_*]'       -->
 33 |   <!-- keyword is discarded, and all bytes until the end of the line    -->
 34 |   <!-- are also discarded.                                              -->
 35 |   <!--                                                                  -->
 36 |   <!-- If the specified header was present several times in the         -->
 37 |   <!-- message, all occurences are concatenated (CRLF seperated)        -->
 38 |   <!-- to be used in place of the '[last_*]' keyword.                   -->
 39 | 
 40 |   <send>
 41 |     <![CDATA[
 42 | 
 43 |       SIP/2.0 180 Ringing
 44 |       [last_Via:]
 45 |       [last_From:]
 46 |       [last_To:];tag=[call_number]
 47 |       [last_Call-ID:]
 48 |       [last_CSeq:]
 49 |       [last_Record-route]
 50 |       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
 51 |       Content-Length: 0
 52 | 
 53 |     ]]>
 54 |   </send>
 55 | 
 56 |   <send retrans="500">
 57 |     <![CDATA[
 58 | 
 59 |       SIP/2.0 200 OK
 60 |       [last_Via:]
 61 |       [last_From:]
 62 |       [last_To:];tag=[call_number]
 63 |       [last_Call-ID:]
 64 |       [last_CSeq:]
 65 |       [last_Record-route]
 66 |       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
 67 |       Content-Type: application/sdp
 68 |       Content-Length: [len]
 69 | 
 70 |       v=0
 71 |       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
 72 |       s=-
 73 |       c=IN IP[media_ip_type] [media_ip]
 74 |       t=0 0
 75 |       m=audio [media_port] RTP/AVP 0
 76 |       a=rtpmap:0 PCMU/8000
 77 | 
 78 |     ]]>
 79 |   </send>
 80 | 
 81 |   <recv request="ACK"
 82 |         optional="true"
 83 |         rtd="true"
 84 |         crlf="true">
 85 |   </recv>
 86 | 
 87 |   <recv request="BYE">
 88 |   </recv>
 89 | 
 90 |   <send>
 91 |     <![CDATA[
 92 | 
 93 |       SIP/2.0 200 OK
 94 |       [last_Via:]
 95 |       [last_From:]
 96 |       [last_To:]
 97 |       [last_Call-ID:]
 98 |       [last_CSeq:]
 99 |       [last_Record-route]
100 |       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
101 |       Content-Length: 0
102 | 
103 |     ]]>
104 |   </send>
105 | 
106 |   <!-- Keep the call open for a while in case the 200 is lost to be     -->
107 |   <!-- able to retransmit it if we receive the BYE again.               -->
108 |   <pause milliseconds="4000"/>
109 | 
110 | 
111 |   <!-- definition of the response time repartition table (unit is ms)   -->
112 |   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
113 | 
114 |   <!-- definition of the call length repartition table (unit is ms)     -->
115 |   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
116 | 
117 | </scenario>
118 | 


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